Over the past several years there has been, and continues to be, a tremendous amount of activity in the area of efficient encoding of speech. For an evolving digital telephone network, a most important application is the possible replacement of the 64,000 bit-per-second (b/s) PCM signal (8 bits per time slot, repeated at an 8 kHz rate) with other coding algorithms for telephony--both in the public switched and private line networks. The reason, of course, is to achieve bandwidth compression.
For a realistic mix of input speech and voiceband data, adaptive differential PCM appears to be a most promising approach. One form of adaptive differential PCM coding is disclosed, for example, in U.S. Pat. No. 4,437,087 issued to D. W. Petr on Mar. 13, 1984 and can be considered a benchmark since a single encoding with this ADPCM coder at 32 kb/s is near to being subjectively equivalent to 64 kb/s .mu.255 PCM. Also see COM XVIII-R31, "Recommendation G.721--32 kbits/s Adaptive Differential Pulse Code Modulation (ADPCM)", VII.sup.th CCITT Plenary Assembly, Maloga-Torremolinos, Spain, Vol. III, pp. 125-159, October 1984, for details of one such prior ADPCM coder and decoder arrangement. This prior ADPCM coder and decoder arrangement employs a so-called locking quantizer (inverse quantizer) adaptation speed control unit which locks the quantizer scale factor for voiceband data and tone like signals, i.e., partial band energy signals.
The prior coder and decoder arrangement although operating satisfactorily in most applications does not operate completely satisfactorily in applications including transitions from a so-called partial band energy signal to another such signal. Partial band energy signals include single tones, multifrequency tones, touch-tones and the like. A specific problem is experienced using frequency shift keyed (FSK) modems. In such modems, a logical 1 is represented by a first single tone signal, e.g., 1200 Hz, and a logical 0 is represented by a second single tone signal, e.g., 2200 Hz. In the continuous carrier mode of operation of a FSK modem, e.g., a 202 type modem, the modem transmits a single frequency tone representing "space", i.e., all logical 1's, between actual characters. A typical signal generated from typing at a keyboard consists of a long interval of the single frequency tone followed by a few bits of data, i.e., characters in ASCII format, then the single frequency tone again and so on.
We have determined that under such signal conditions the prior ADPCM coder and decoder arrangement disclosed in the D. W. Petr patent and the CCITT Recommendation G.721, both cited above, has problems in tracking the transitions from one tone to another, which results in the generation of large amounts of impulse noise at the output of the decoder. The impulse noise causes character errors between transmitting and receiving modems.
Specifically, the impulse noise is caused by the adaptive predictor and adaptive quantizer in the coder and by the adaptive predictor and adaptive inverse quantizer in the decoder adapting almost perfectly to the single frequency tone, e.g., 1200 Hz. That is to say, the adaptive predictor is generating a signal estimate which is substantially identical to the single frequency tone and the adaptive quantizer (inverse quantizer) is locking and scaling down to the very small difference signal resulting from the algebraic subtraction of the signal estimate from the single frequency tone. Upon a transition to another single frequency tone, e.g., 2200 Hz, the difference signal can become extremely large; sometimes more than twice the magnitude of the single frequency tone input signal. Consequently, the decoder output does not reflect the coder input signal for many samples until the predictor and quantizer (inverse quantizer) have adapted to the new single frequency tone input signal.